Can you explain where i have to configure it on freepbx?
I have configured outgoing mail address on fax configuration on settings TAB
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FreePBX 15 Voip - Fax - Chan PJSIP issue
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FreePBX 15 Voip - Fax - Chan PJSIP issue
Are you getting other mail from the PBX (voicemail messages, update notifications, etc.)?
What does
mailq
show?
Can you manually send mail?, e.g.
mail -s test you@yourdomain.com
.
(after typing the mail command, type just a . on the next line).
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Polycom - Valet Park Retrieval quit working with EPM Update
Ok - I am trying to use Sangoma Connect and it told me I had to update EPM to have Connect work correctly - I was on 15.0.24.something (didn’t write it down) and it made me upgrade to 15.0.39.49 - Now Valet-parking quit working - You can park the call when you press the spot BLF-XFER button (automata) but when you try and retrieve it, it acts like you are trying to dial out - WHY???
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Sometimes caller can't hear me
Also check that your router/firewall is set to forward the RTP port range (default is UDP 10000-20000) to the PBX.
If you still have trouble, capture traffic with tcpdump to see whether RTP is going out to the correct IP address and port.
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Asterisk locks up not allowing any calls in or out. (even between internal extensions)
I made this change and it hasn’t dropped since.
Mar '19
Hi,
I had this exact same problem with PJSIP and it is related to this Asterisk bug: https://issues.asterisk.org/jira/browse/ASTERISK-27821
Changing all PJSIP extension’s MWI Subscription Type from Auto to Solicited solves the problem.
Fraser.
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Polycom - Valet Park Retrieval quit working with EPM Update
EPM as a paid commercial module includes support. Open a ticket with details:
https://wiki.sangoma.com/display/FPAS/How+To+Open+A+Support+Ticket
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EndPoint Manager pointing Grandstream to wrong port for idle screen xml
I am provisioning my Grandstream phones via TFTP on port 69. The generated config files for the phones point to the correct port for provisioning and firmware but for the “Idle Screen XML Server Path” they are incorrectly pointing to port 83.
# Enable Idle Screen XML Download. 0-No, 1-YES HTTP, 2-YES TFTP
P340 = 2
# Idle Screen XML Server Path
P341 = 10.0.0.6:83/grandstream-ccg_grandstream-GXP-2100.xml
I am running EndPoint Manager 14.0.68.48
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Polycom - Valet Park Retrieval quit working with EPM Update
Submitted ticket - Will post resolution here when I get one!
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Telekom Germany SIP Trunk SRV Records
Yes. You got it right. It’s best to provide a complete RPZ (as it is called by Bind) and provide Asterisk a reduced answer to the DNS lookups. This could look like this (a script periodically creates the following code):
server $yourBindIP
zone rpz-tonline
update delete tel.t-online.de.rpz-tonline.
update delete _sips._tcp.tel.t-online.de.rpz-tonline.
update delete _sip._tcp.tel.t-online.de.rpz-tonline.
update add tel.t-online.de.rpz-tonline. 60 NAPTR 10 0 "s" "SIPS+D2T" "" _sips._tcp.tel.t-online.de.
update add tel.t-online.de.rpz-tonline. 60 NAPTR 30 0 "s" "SIP+D2T" "" _sip._tcp.tel.t-online.de.
update add _sips._tcp.tel.t-online.de.rpz-tonline. 60 SRV 10 0 5061 s-eps-110.edns.t-ipnet.de.
update add _sip._tcp.tel.t-online.de.rpz-tonline. 60 SRV 10 0 5060 s-epp-110.edns.t-ipnet.de.
sendwhich can be send to Bind with nsupdate.
It results in
$ORIGIN .
$TTL 660 ; 11 minutes
rpz-tonline IN SOA localhost. root.localhost. (
51 ; serial
10800 ; refresh (3 hours)
3600 ; retry (1 hour)
604800 ; expire (1 week)
3600 ; minimum (1 hour)
)
NS localhost.
$ORIGIN rpz-tonline.
$TTL 60 ; 1 minute
tel.t-online.de NAPTR 30 0 "s" "SIP+D2T" "" _sip._tcp.tel.t-online.de.
NAPTR 10 0 "s" "SIPS+D2T" "" _sips._tcp.tel.t-online.de.
$ORIGIN _tcp.tel.t-online.de.rpz-tonline.
_sip SRV 10 0 5060 s-epp-110.edns.t-ipnet.de.
_sips SRV 10 0 5061 s-eps-110.edns.t-ipnet.de.named.conf
options {
...
response-policy {
zone "rpz-tonline";
};
}
zone "rpz-tonline" {
type master;
file "/var/named/rpz-tonline-override";
allow-query { any; };
allow-transfer { any; };
allow-update { any; };
};The final destination servers (*.t-ipnet.de) are resolved directly again via DNS (they provide just one answer, which is fine). You have to take care if one of the SRV results above you’re using changes / disappears. If this happens, you have to unregister, update the DNS and register again. To be sure to not break any ongoing call, you have to wait until no call is active (that can be done in the script, too, via a CLI call checking for running calls to the trunk).
@sorvani: German Telekom provides 3 different servers (I think these are completely different and solely acting clusters) and expects that you register to one of them and all further requests must go exactly to this server you initially registered to - requests to any other server / cluster is denied (they don’t know you). Asterisk can’t handle this at the moment.
There are situations where a registration fails (mostly on startup - if you register more than one number). Asterisk then switches to the second server (and proceeds) and afterwards uses for an outgoing call (or options-request) the first server again. All those requests are failing because they get access denied, because you’re registered to another cluster.
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Sangoma Dect Phones OpenVPN?
Is it possible to set up sangoma dect phones or base station to use the FreePBX OpenVPN like you can with the regular desk phones in EPM?
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FreePBX 15 Voip - Fax - Chan PJSIP issue
Ok, postfix not configured well but now i have another issue. I select pdf type for files of fax but i have received tif type…why?
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How to change the licence duration for a commercial product?
Hello,
I went and try to buy a licence on the portal but it asks for “Deployment Option” and I dont know what to enter
I was expecting that I would receive a licence number, so I typed licence and that was “not found”.
Any hint ?
Thanks
Gilbert
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iPhone will not connect only shows registering
There are two parts to this, provision which you can lockdown the ports to the Sangoma IP’s and then registration, which if you are using your Cell network means you are unlikely to have static IPs. So you will need to open your SIP port to the outside world, which if you enable responsive firewall will be fine.
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Saturation in lines when there are many calls
I have an apparently well-sized machine:
But when the queued calls are greater than 30 … the problems start making calls.
How can I check that the machine is properly dimensioned?
Could it be something else’s problem?
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System restarted due to power interruption then it stuck on boot
Hi, I’ve had this problem for a long time and till now can’t resolve it. Did some google search and tried the suggested solutions (fsck-no error found) but still stuck on boot. Managed to create a backup using clonezilla and restored in the virtualbox, trying to resolve it from here.
I’m still newbie with linux but can work around, any help would be great. Thank You
stuck here
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FreePBX in Cloud and Gateway FXO
Hello to all,
I state that about the subject, I have some fundamental knowledge. But the time has come to address them so that I can solve the needs I am having.
Before describing the topic and problems for which I wrote this post, I make a quick summary of my current system (FreePBX) and what I would like to add.
Synthetic summary.
- My FreePBX is installed in the Cloud (it can be reached through a static public address made available by my data center provider).
In FreePBX, some sip trunks are currently configured to connect some geographic numbers purchased from VoIP line providers.
-
My internet connectivity and PSTN line arrives in my office through the router provided by my telephone and data service provider. Also, in my office on my desk, I have a VoIP phone with some extensions registered. Everything works fine.
-
Now, I would like to be able to use the VoIP numbers also my PSTN line.
To fulfill my desire to make both inbound and outbound calls with my PSTN line, I purchased a VoIP FXO Gateway (Yeastar TA810). The FXO VoIP Gateway is placed inside my office network, which is assigned a static IP address and I connect one of the two RJ11 telephone sockets on my router to an FXO port on my Gateway.
In the Gateway I create a Service Provider trunk that points to my FreePBX in the Cloud through its public static address:
I also provide in LAN settings in my Gateway to set a static private address (consistent with the local network of my office as follows):
In the SIP settings of my Gateway I have configured as follows:
As you can see, I have set UDP as port 6060 because when FreePBX tries to connect to my router’s static public address in the office, the router does not allow port forwarding to port 5060 to the Gateway FXO placed in the internal office network. While if I use the 6060 port, my router will be able to do the port forwarding of the 6060 UDP port towards the Gateway in the corporate LAN.
Finally, to manage outbound calls from the VoIP phone in the Gateway, set the IP-> Port in the Routes Settings section as follows:
While managing incoming calls, always in the Routes Settings section, I set the Port-> Ip / Port as follows:
In FreePBX placed in the Cloud (external network to that of the office) I configured a pjsip trunk as follows:
I enable this trunk (for the management of my PSTN). On my VoIP phone placed on the desk, all the configured extensions are disconnected, including those that use the VoIP trunks. Also is disconnected the extension dedicated to using the PSTN.
What is not understandable to me is that if I use a softphone on my mobile phone, using 4G data connection(Zoiper), all extensions work, I can make outgoing calls and receive calls PSTN line placed in my office.
Could you help me understand what I’m doing wrong? Even simply from a conceptual point of view? I wonder if an FXO Gateway must be inside the same LAN as the PBX, but in this case, it is a virtualized FreePBX in the Cloud… Why does everything work with the mobile phone using the 4G connection? Of course, if I use my mobile phone with the office Wi-FI, it stops working too.
I apologize for the long post. Any help and advice would be valuable to me.
I appreciate any help you can provide.
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Wrong time in SYSTEM ADMIN
hi there
SYSYTEM ADMIN time not switching to EEST from EET
(it is in EEST in CLI)
FreePBX 13.0.197.28
System Admin 13.0.95.4
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Wrong time in SYSTEM ADMIN
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System restarted due to power interruption then it stuck on boot
Hello @Verto15,
You have a duplicated record in your queue table. Do you use a 3rd party module for managing the queues?
Do you have access to the server via SSH? If yes, you need to access this database and delete this record from the table.
What is the Asterisk/Freepbx version that you are using?
Thank you,
Daniel Friedman
Trixton LTD.
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Sangoma H20 Headset PC audio problem
I’ve tested Sangoma H20 on two PCs and both face the same problem.
Looks like the headset goes in a powersave mode when no audio is received, but sometimes happens that the base can’t wake up the headphone, so call is processed because Zulu finds the base connected, but there’s no audio between base and headphone.
Tried different audio configuration in both Zulu and Win10 settings but still happens when doing two or more calls in a row.
What can I do?
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